Recently, with the advances in computers, networking and communications streaming audio contents over networks such as the Internet, wireless local area networks, home networks and commercial cellular phone systems is becoming a mainstream means of audio service delivery. It is believed that with the progress of the broadband network infrastructures, including xDSL, fiber optics, and broadband wireless access, bit rates for these channels are quickly approaching those for delivering high sampling-rate, high amplitude resolution (e.g. 96 kHz, 24 bit/sample) lossless audio signals. On the other hand, there are still application areas where high-compression digital audio formats, such as MPEG-4 AAC (described in [1]) are required. As a result, interoperable solutions that bridge the current channels and the rapidly emerging broadband channels are highly demanded. In addition, even when broadband channels are widely available and the bandwidth constraint is ultimately removed, a bit-rate-scalable coding system that is capable to produce a hierarchical bit-stream whose bit-rates can be dynamically changed during transmission is still highly favorable. For example, for applications where packet loss occurs occasionally due to accidents or resource sharing requirements, the current broadband waveform representations such as PCM (Pulse Code Modulation) and lossless coding formats may suffer serious distortions in a streaming situation. However, this problem can be solved if one could set packet priorities in the case that network resources are dynamically changing. Finally, a bit-rate-scalable coding system also provides the server advantageous for audio streaming services, where graceful QoS degradation could be achieved if an excessive number of demands from client sites come.
Previously many lossless audio coding algorithms have been proposed (see [2]-[8]). Most approaches rely on a prediction filter to remove the redundancy of the original audio signals while the residuals are entropy coded (as described in [5]-[12]). Due to the existence of the predictive filters, the bit-streams generated by these prediction based approaches are difficult and not efficient (see [5],[6]), if not impossible, to be scaled to achieve bit-rate scalability. Other approaches, such as described in [3], build the lossless audio coder through a two layer approach where the original audio signals are first coded with a lossy encoder and its residual is then lossless coded with a residual encoder. Although this two layer design provides some sort of bit-rate scalability, its granularity is too coarse to be appreciated by audio streaming applications. Audio codecs that provide the fine grain scalability on bit-rate were previously proposed in [4] and [18], however, unlike the system to be discussed here, those codecs don't provide the backward compatibility that the lossy bit-streams produced by both codecs are incompatible to any existing audio codec.
In [21], [22], [23] perceptual models are described.
The object of the invention is to provide a method for encoding a digital signal in a scalable bitstream wherein backward compatibility can be maintained.